Digital transmission systems are known wherein real time signals (e.g voice) are transmitted in packet form. An advantage of packet switching derives from it enabling easily mixing real time and data. However the protocols are completely different. Since for real time signals end-to-end delays are constrained and real time is required for packets restitution, then erroneous packets, and packets which have been too much delayed waiting in network node's queues must be discarded and cannot be retransmitted. This results in packet losses. While data packets may be delayed and retransmitted if required.
In the following description, real time signal will be considered as relating to voice signal, but the invention applies to other real time signals, to video images for instance.
One has today efficient methods to reconstruct, at the receiving end, some of the lost voice packets (see European Applications 0,139,803 and 0,162,173 to the same applicant) which generate no audible degradation up to a loss of 5%, keep intelligibility and speaker recognition ability up to a loss of 33% with a graceful degradation of the speech quality. But unfortunately, these methods are not efficient as far as voice quality is concerned if two packets or more are consecutively lost for a given speaker.
One object of this invention is to provide an efficient and simple method to drastically decrease consecutive voice packets losses, in a digital network.
Another object is to provide means for freeing bandwidth for data traffic in a voice packet network.